subject:"\[asterisk\-users\] Asterisk 13 PJSIP with Snom 710"

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

2016-09-09 Thread Madushan Geethanga

thanks for the reply. if i config the extension in softphone it works fine.
but with snom its not working

Bet Regards,
Madushan

On Fri, Sep 9, 2016 at 10:31 PM, Madushan Geethanga  wrote:

> yes I have unchecked it.
>
> On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI 
> wrote:
>
>> Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
>>
>>> Hi,
>>>
>>
>> If you're not using RTP encryption did you uncheck the option in your RTP
>> TAB from identity ?
>>
>>
>>> This is the log. ex dialling 0 from snom phone
>>>
>>>
>>> <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
>>>  --->
>>> INVITE sip:0@54.206.59.252 ;user=phone
>>> SIP/2.0
>>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>>> From: "outburns00-nhvg5vjjn6-2001"
>>> >> >;tag=1bb809zgaa
>>> To: mailto:sip%3A0@54.206.59.252>;user=phone>
>>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>>> CSeq: 1 INVITE
>>> Max-Forwards: 70
>>> User-Agent: snom710/8.7.5.35 
>>> Contact: >> >;reg-id=1
>>>
>>> X-Serialnumber: 000413747C96
>>> P-Key-Flags: resolution="31x13", keys="4"
>>> Accept: application/sdp
>>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
>>> PRACK, MESSAGE, INFO, UPDATE
>>> Allow-Events: talk, hold, refer, call-info
>>> Supported: timer, 100rel, replaces, from-change
>>> Session-Expires: 3600
>>> Min-SE: 90
>>> Content-Type: application/sdp
>>> Content-Length: 405
>>>
>>> v=0
>>> o=root 2136927789 2136927789 IN IP4 192.168.2.28
>>> s=call
>>> c=IN IP4 123.231.72.210
>>> t=0 0
>>> m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:99 G726-32/8000
>>> a=rtpmap:112 AAL2-G726-32/8000
>>> a=rtpmap:18 G729/8000
>>> a=fmtp:18 annexb=no
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> <--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
>>>  --->
>>> SIP/2.0 401 Unauthorized
>>> Via: SIP/2.0/UDP
>>> 123.231.72.210:45835;rport=33878;received=123.231.72.210;bra
>>> nch=z9hG4bK-bskkkx1t5bas
>>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>>> From: "outburns00-nhvg5vjjn6-2001"
>>> >> >;tag=1bb809zgaa
>>> To: >> ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>>> CSeq: 1 INVITE
>>> WWW-Authenticate: Digest
>>> realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0c
>>> cea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
>>> Server: Asterisk PBX certified/13.8-cert2
>>> Content-Length:  0
>>>
>>>
>>> <--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
>>>  --->
>>> ACK sip:0@54.206.59.252 ;user=phone
>>> SIP/2.0
>>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>>> From: "outburns00-nhvg5vjjn6-2001"
>>> >> >;tag=1bb809zgaa
>>> To: >> ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>>> CSeq: 1 ACK
>>> Max-Forwards: 70
>>> User-Agent: snom710/8.7.5.35 
>>> Contact: >> >;reg-id=1
>>> Content-Length: 0
>>>
>>>
>>> Best Regards,
>>> Madushan
>>>
>>>
>>>
>>> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
>>> mailto:mgliyanage...@gmail.com>> wrote:
>>>
>>> Hi,
>>>
>>> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
>>> inbound is working fine but i cannot dial out. i don't hear anything
>>> on the phone and asterisk CLI also does not show anything. my config
>>> is. please advice.
>>>
>>> [2001]
>>> type=endpoint
>>> context=out-local
>>> disallow=all
>>> allow=ulaw
>>> allow=alaw
>>> transport=system-udp
>>> auth=2001
>>> aors=2001
>>> direct_media=no
>>> rtp_symmetric=yes
>>> force_rport=yes
>>> allow=alaw
>>> allow=speex
>>> allow=speex16
>>> allow=speex32
>>> allow=gsm
>>>
>>>
>>> [2001]
>>> type=aor
>>> qualify_frequency=5000
>>> authenticate_qualify=yes
>>> max_contacts=1
>>> remove_existing=yes
>>>
>>> [2001]
>>> type=auth
>>> auth_type=userpass
>>> password=test
>>> username=test
>>>
>>> Best Regards,
>>> Madushan
>>>
>>>
>>>
>>>
>>>
>> --
>> _
>> -- Bandwidth and Colocation Provi

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

2016-09-09 Thread Madushan Geethanga

yes I have unchecked it.

On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI 
wrote:

> Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
>
>> Hi,
>>
>
> If you're not using RTP encryption did you uncheck the option in your RTP
> TAB from identity ?
>
>
>> This is the log. ex dialling 0 from snom phone
>>
>>
>> <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
>>  --->
>> INVITE sip:0@54.206.59.252 ;user=phone
>> SIP/2.0
>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>> From: "outburns00-nhvg5vjjn6-2001"
>> > >;tag=1bb809zgaa
>> To: mailto:sip%3A0@54.206.59.252>;user=phone>
>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>> CSeq: 1 INVITE
>> Max-Forwards: 70
>> User-Agent: snom710/8.7.5.35 
>> Contact: > >;reg-id=1
>>
>> X-Serialnumber: 000413747C96
>> P-Key-Flags: resolution="31x13", keys="4"
>> Accept: application/sdp
>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
>> PRACK, MESSAGE, INFO, UPDATE
>> Allow-Events: talk, hold, refer, call-info
>> Supported: timer, 100rel, replaces, from-change
>> Session-Expires: 3600
>> Min-SE: 90
>> Content-Type: application/sdp
>> Content-Length: 405
>>
>> v=0
>> o=root 2136927789 2136927789 IN IP4 192.168.2.28
>> s=call
>> c=IN IP4 123.231.72.210
>> t=0 0
>> m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:99 G726-32/8000
>> a=rtpmap:112 AAL2-G726-32/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>> a=sendrecv
>>
>> <--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
>>  --->
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 123.231.72.210:45835;rport=33878;received=123.231.72.210;bra
>> nch=z9hG4bK-bskkkx1t5bas
>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>> From: "outburns00-nhvg5vjjn6-2001"
>> > >;tag=1bb809zgaa
>> To: > ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>> CSeq: 1 INVITE
>> WWW-Authenticate: Digest
>> realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0c
>> cea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
>> Server: Asterisk PBX certified/13.8-cert2
>> Content-Length:  0
>>
>>
>> <--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
>>  --->
>> ACK sip:0@54.206.59.252 ;user=phone SIP/2.0
>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
>> From: "outburns00-nhvg5vjjn6-2001"
>> > >;tag=1bb809zgaa
>> To: > ;user=phone>;tag=z9hG4bK-bskkkx1t5bas
>> Call-ID: 313437333433383639323238313539-ahn3begiq66q
>> CSeq: 1 ACK
>> Max-Forwards: 70
>> User-Agent: snom710/8.7.5.35 
>> Contact: > >;reg-id=1
>> Content-Length: 0
>>
>>
>> Best Regards,
>> Madushan
>>
>>
>>
>> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
>> mailto:mgliyanage...@gmail.com>> wrote:
>>
>> Hi,
>>
>> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
>> inbound is working fine but i cannot dial out. i don't hear anything
>> on the phone and asterisk CLI also does not show anything. my config
>> is. please advice.
>>
>> [2001]
>> type=endpoint
>> context=out-local
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> transport=system-udp
>> auth=2001
>> aors=2001
>> direct_media=no
>> rtp_symmetric=yes
>> force_rport=yes
>> allow=alaw
>> allow=speex
>> allow=speex16
>> allow=speex32
>> allow=gsm
>>
>>
>> [2001]
>> type=aor
>> qualify_frequency=5000
>> authenticate_qualify=yes
>> max_contacts=1
>> remove_existing=yes
>>
>> [2001]
>> type=auth
>> auth_type=userpass
>> password=test
>> username=test
>>
>> Best Regards,
>> Madushan
>>
>>
>>
>>
>>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   h

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

2016-09-09 Thread Administrator TOOTAI


Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :

Hi,


If you're not using RTP encryption did you uncheck the option in your 
RTP TAB from identity ?




This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
 --->
INVITE sip:0@54.206.59.252 ;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001"
mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: mailto:sip%3A0@54.206.59.252>;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 
Contact: http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
X-Serialnumber: 000413747C96
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Content-Type: application/sdp
Content-Length: 405

v=0
o=root 2136927789 2136927789 IN IP4 192.168.2.28
s=call
c=IN IP4 123.231.72.210
t=0 0
m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
123.231.72.210:45835;rport=33878;received=123.231.72.210;branch=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
From: "outburns00-nhvg5vjjn6-2001"
mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: mailto:sip%3A0@54.206.59.252>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
CSeq: 1 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/13.8-cert2
Content-Length:  0


<--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
 --->
ACK sip:0@54.206.59.252 ;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001"
mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: mailto:sip%3A0@54.206.59.252>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 
Contact: http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
mailto:mgliyanage...@gmail.com>> wrote:

Hi,

I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
inbound is working fine but i cannot dial out. i don't hear anything
on the phone and asterisk CLI also does not show anything. my config
is. please advice.

[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm


[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards,
Madushan






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

2016-09-09 Thread Madushan Geethanga

Hi,

This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 --->
INVITE sip:0@54.206.59.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001" <
sip:outburns00-nhvg5vjjn6-2001@54.206.59.252>;tag=1bb809zgaa
To: 
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: ;reg-id=1
X-Serialnumber: 000413747C96
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Content-Type: application/sdp
Content-Length: 405

v=0
o=root 2136927789 2136927789 IN IP4 192.168.2.28
s=call
c=IN IP4 123.231.72.210
t=0 0
m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 123.231.72.210:45835
;rport=33878;received=123.231.72.210;branch=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
From: "outburns00-nhvg5vjjn6-2001" <
sip:outburns00-nhvg5vjjn6-2001@54.206.59.252>;tag=1bb809zgaa
To: ;tag=z9hG4bK-bskkkx1t5bas
CSeq: 1 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/13.8-cert2
Content-Length:  0


<--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878 --->
ACK sip:0@54.206.59.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001" <
sip:outburns00-nhvg5vjjn6-2001@54.206.59.252>;tag=1bb809zgaa
To: ;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35
Contact: ;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga 
wrote:

> Hi,
>
> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
> working fine but i cannot dial out. i don't hear anything on the phone and
> asterisk CLI also does not show anything. my config is. please advice.
>
> [2001]
> type=endpoint
> context=out-local
> disallow=all
> allow=ulaw
> allow=alaw
> transport=system-udp
> auth=2001
> aors=2001
> direct_media=no
> rtp_symmetric=yes
> force_rport=yes
> allow=alaw
> allow=speex
> allow=speex16
> allow=speex32
> allow=gsm
>
>
> [2001]
> type=aor
> qualify_frequency=5000
> authenticate_qualify=yes
> max_contacts=1
> remove_existing=yes
>
> [2001]
> type=auth
> auth_type=userpass
> password=test
> username=test
>
> Best Regards,
> Madushan
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13 PJSIP with Snom 710

2016-09-09 Thread Madushan Geethanga

Hi,

I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
working fine but i cannot dial out. i don't hear anything on the phone and
asterisk CLI also does not show anything. my config is. please advice.

[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm


[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards,
Madushan
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

[asterisk-users] Asterisk 13 PJSIP with Snom 710

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